Apparatus for processing an audio signal and method thereof

ABSTRACT

A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal.

CROSS-REFERENCE TO RELATED APPLICATION

This application claims the benefit of U.S. Provisional Application No.61/158,388 filed on Mar. 8, 2009 and U.S. Provisional Application No.61/164,459, filed on Mar. 29, 2009, which are hereby incorporated byreference.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to an apparatus for processing an audiosignal and method thereof. Although the present invention is suitablefor a wide scope of applications, it is particularly suitable forprocessing an audio signal.

2. Discussion of the Related Art

Generally, an audio signal is outputted via a loud speaker provided to atelevision set, a portable device or the like or a headset and the like.Before the audio signal is outputted via a speaker or the like, an audioprocessor can perform such processing as noise canceling, normalizing,volume adjusting and the like on the audio signal.

However, according to a related art, such processing as volume control,bass control and the like is performed individually and independently.Since sound intensity based on a sound pressure level is considered inperforming the volume control, it may cause a problem that loudness ofthe sound actually falling on human ears makes a differenceunintentionally.

Moreover, in performing the bass control, if a frequency response of aloud speaker is low for a low frequency or bass is excessively boosted,it may cause a problem that a signal is distorted.

SUMMARY OF THE INVENTION

Accordingly, the present invention is directed to an apparatus forprocessing an audio signal and method thereof that substantially obviateone or more of the problems due to limitations and disadvantages of therelated art.

An object of the present invention is to provide an apparatus forprocessing an audio signal and method thereof, by which a volume can becontrolled using loudness of sound actually audible to a human for eachsound pressure level.

Another object of the present invention is to provide an apparatus forprocessing an audio signal and method thereof, by which a volume can becontrolled per frequency band using an absolute threshold of hearing. Inthis case, the absolute threshold of hearing is a minimumhuman-recognizable sound pressure level (SPL)

Another object of the present invention is to provide an apparatus forprocessing an audio signal and method thereof, by which a volume can becontrolled in a manner of applying both a linear gain for adjusting again in proportion to a user input and a non-linear gain for adjusting again on the basis of an absolute threshold of hearing.

A further object of the present invention is to provide an apparatus forprocessing an audio signal and method thereof, by which an adaptivefilter can be applied to improve a bass if a frequency response of aloudspeaker is low.

Additional features and advantages of the invention will be set forth inthe description which follows, and in part will be apparent from thedescription, or may be learned by practice of the invention. Theobjectives and other advantages of the invention will be realized andattained by the structure particularly pointed out in the writtendescription and claims thereof as well as the appended drawings.

To achieve these and other advantages and in accordance with the purposeof the present invention, as embodied and broadly described, a methodfor processing an audio signal, comprising: receiving, by an audioprocessing apparatus, an input signal; receiving user gain input;generating a linear gain factor and a non-linear gain factor using theuser gain input; modifying the non-linear gain factor using absolutethreshold of hearing and power of the input signal to generate amodified non-linear gain factor; and, applying the linear gain factorand the modified non-linear gain factor to the audio signal is provided.

According to the present invention, the non-linear gain factor and thelinear gain factor are generated based on whether the user gain input islower or higher than at least one of a low reference value and a highreference value.

According to the present invention, when the user gain input is lowerthan a low reference value, the non-linear gain factor is generated as afixed value, and the linear gain factor is generated using the user gaininput, and, when the user gain input is equal to or higher than a lowreference value, the non-linear gain factor is generated using the usergain input and the linear gain factor is generated as a fixed value.

According to the present invention, the fixed value is determinedaccording to at least one of the low reference value and a highreference value.

According to the present invention, when the user gain input is equal orhigher than a high reference value, the non-linear gain factor is equalto one and the linear gain factor is equal to the user gain input.

According to the present invention, when the user gain input is equal orhigher than a low reference value, the non-linear gain factor is equalto the user gain input and the linear gain factor is equal to one.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, an apparatus for processing an audiosignal, comprising: a receiving unit receiving an input signal; a usergain receiving part receiving user gain input; a gain splitting partgenerating a linear gain factor and a non-linear gain factor using theuser gain input; a non-linear gain modifying part modifying thenon-linear gain factor using absolute threshold of hearing and power ofthe input signal to generate a modified non-linear gain factor; and, again applying part applying the linear gain factor and the modifiednon-linear gain factor to the audio signal is provided.

According to the present invention, the non-linear gain factor and thelinear gain factor are generated based on whether the user gain input islower or higher than at least one of a low reference value and a highreference value.

According to the present invention, when the user gain input is lowerthan a low reference value, the non-linear gain factor is generated as afixed value, and the linear gain factor is generated using the user gaininput, and, when the user gain input is equal to or higher than a lowreference value, the non-linear gain factor is generated using the usergain input, and the linear gain factor is generated as a fixed value.

According to the present invention, the fixed value is determinedaccording to at least one of the low reference value and a highreference value.

According to the present invention, when the user gain input is equal orhigher than a high reference value, the non-linear gain factor is equalto one and the linear gain factor is equal to the user gain input.

According to the present invention, when the user gain input is equal orhigher than a low reference value, the non-linear gain factor is equalto the user gain input and the linear gain factor is equal to one.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, a computer-readable medium havinginstructions stored thereon, which, when executed by a processor, causesthe processor to perform operations, comprising: receiving, by an audioprocessing apparatus, an input signal; receiving user gain input;generating a linear gain factor and a non-linear gain factor using theuser gain input; modifying the non-linear gain factor using absolutethreshold of hearing and power of the input signal to generate amodified non-linear gain factor; and, applying the linear gain factorand the modified non-linear gain factor to the audio signal is provided.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, an apparatus for processing an audiosignal, a method for processing an audio signal, comprising: receiving,by an audio processing apparatus, an input signal; estimating indicatorfunction using a signal power of the input signal; obtaining an adaptedfilter using the indicator function and an equalization filter; and,generating an output signal by applying the adapted filter to the inputsignal is provided.

According to the present invention, the indicator function is forheadroom for a loudspeaker.

According to the present invention, the indicator function is estimatedfurther using a maximum power and a minimum power.

According to the present invention, when the signal power is greaterthan the maximum power, the adapted filter corresponds to a unityfilter, when the signal power is equal to or smaller than the maximumpower and greater than the minimum power, the adapted filter correspondsto a combination of the unity filter and the equalization filter, andwhen the signal power is equal to or smaller than the minimum power, theadapted filter corresponds to the equalization filter.

According to the present invention, the signal power is generated usinga weight filter and the input signal.

According to the present invention, the indicator function ranges fromzero to one.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, an apparatus for processing an audiosignal, comprising: a receiving part receiving, by an audio processingapparatus, an input signal; an indicator function estimating partestimating indicator function using a signal power of the input signal;a filter obtain part obtaining an adapted filter using the indicatorfunction and an equalization filter; and, a filter applying partgenerating an output signal by applying the adapted filter to the inputsignal is provided.

According to the present invention, the indicator function is estimatedfurther using headroom for a loudspeaker.

According to the present invention, the indicator function is estimatedfurther using a maximum power and a minimum power.

According to the present invention, when the signal power is greaterthan the maximum power, the adapted filter corresponds to a unityfilter, when the signal power is equal to or smaller than the maximumpower and greater than the minimum power, the adapted filter correspondsto a combination of the unity filter and the equalization filter, andwhen the signal power is equal to or smaller than the minimum power, theadapted filter corresponds to the equalization filter.

According to the present invention, the signal power is generated usinga weight filter and the input signal.

According to the present invention, the indicator function ranges fromzero to one.

To further achieve these and other advantages and in accordance with thepurpose of the present invention, a computer-readable medium havinginstructions stored thereon, which, when executed by a processor, causesthe processor to perform operations, comprising: receiving, by an audioprocessing apparatus, an input signal; estimating indicator functionusing a signal power of the input signal; obtaining an adapted filterusing the indicator function and an equalization filter; and, generatingan output signal by applying the adapted filter to the input signal isprovided.

It is to be understood that both the foregoing general description andthe following detailed description are exemplary and explanatory and areintended to provide further explanation of the invention as claimed.

BRIEF DESCRIPTION OF THE DRAWINGS

The accompanying drawings, which are included to provide a furtherunderstanding of the invention and are incorporated in and constitute apart of this specification, illustrate embodiments of the invention andtogether with the description serve to explain the principles of theinvention.

In the drawings:

FIG. 1 is a block diagram of an audio signal processing apparatusaccording to an embodiment of the present invention;

FIG. 2 is a block diagram for details of a volume control unit accordingto an embodiment of the present invention;

FIG. 3 is a graph for an example of an absolute threshold of hearing;

FIG. 4 is a graph for an example of an absolute threshold of hearing insubband power domain;

FIG. 5 is a detailed block diagram of a gain splitting unit shown inFIG. 2 according to a first embodiment of the present invention;

FIG. 6 is a flowchart for a gain generating method according to thefirst embodiment shown in FIG. 5;

FIG. 7 is a diagram for per-interval variations of linear and non-lineargains generated by the gain splitting unit shown in FIG. 6;

FIG. 8 is a detailed block diagram of a gain splitting unit shown inFIG. 2 according to a second embodiment of the present invention;

FIG. 9 is a flowchart for a gain generating method according to thesecond embodiment shown in FIG. 8;

FIG. 10 is a diagram for per-interval variations of linear andnon-linear gains generated by the gain splitting unit shown in FIG. 8;

FIG. 11 is a detailed block diagram for an embodiment of a bass controlunit according to an embodiment of the present invention;

FIG. 12 is a flowchart of a method of controlling bass according to anembodiment of the present invention;

FIG. 13 is a graph for examples of a loudspeaker response and a targetequalization filter;

FIG. 14 is a graph for an indicator function and a variation of anadaptive filter according to a signal power;

FIG. 15 is a graph for a loudspeaker response according to an indicatorfunction;

FIG. 16 is a schematic block diagram of a product in which an audiosignal processing apparatus according to one embodiment of the presentinvention is implemented; and

FIG. 17 is a diagram for explaining relations between products in whichan audio signal processing apparatus according to one embodiment of thepresent invention is implemented.

DETAILED DESCRIPTION OF THE INVENTION

Reference will now be made in detail to the preferred embodiments of thepresent invention, examples of which are illustrated in the accompanyingdrawings. First of all, terminologies or words used in thisspecification and claims are not construed as limited to the general ordictionary meanings and should be construed as the meanings and conceptsmatching the technical idea of the present invention based on theprinciple that an inventor is able to appropriately define the conceptsof the terminologies to describe the inventor's invention in best way.The embodiment disclosed in this disclosure and configurations shown inthe accompanying drawings are just one preferred embodiment and do notrepresent all technical idea of the present invention. Therefore, it isunderstood that the present invention covers the modifications andvariations of this invention provided they come within the scope of theappended claims and their equivalents at the timing point of filing thisapplication.

According to the present invention, terminologies not disclosed in thisspecification can be construed as the following meanings and conceptsmatching the technical idea of the present invention. Specifically,‘information’ in this disclosure is the terminology that generallyincludes values, parameters, coefficients, elements and the like and itsmeaning can be construed as different occasionally, by which the presentinvention is non-limited.

In this disclosure, an audio signal indicates a signal identifiable viaan auditory sense to be discriminated from a video signal in a broadsense. In a narrow sense, the audio signal is a signal having no speechproperty or a less speech property to be discriminated from a speechsignal. According to the present invention, an audio signal needs to beinterpreted in a broad sense but can be understandable as a narrow-senseaudio signal in case of being discriminated from a speech signal.

FIG. 1 is a block diagram of an audio signal processing apparatusaccording to an embodiment of the present invention.

Referring to FIG. 1, an audio signal processing apparatus according toan embodiment of the present invention includes at least one of a volumecontrolling unit 300 and a bass controlling unit 400 and is able tofurther include a noise canceling unit 100 and a normalizing unit 200.

The noise canceling unit 100 detects or estimates the noise contained inan audio signal for an input audio signal y[n]. Based on the containedextent of the noise, the noise canceling unit 100 determines a gain pertime or frequency and then applies the determined gain to the audiosignal.

The normalizing unit 200 controls a dynamic range of an input signal bynormalizing the input audio signal y[n] or a signal through the noisecanceling unit 100 based on a user gain input (g). Further, thenormalizing unit 200 is able to adaptively adjust a maximum dynamicrange and a local dynamic range of an input signal. In this case, theuser gain input is information inputted by a ser or device settinginformation and may include a value corresponding to a full-band gain(e.g., 10 dB irrespective of a frequency band).

The volume controlling unit 300 controls a volume for thenoise-cancelled signal, the normalized signal s(n) or the input signaly[n] based on the user gain input (g). In doing so, it is particularlyable to further use an absolute hearing threshold P_(H) in a manner ofmodifying the user gain input (g) and then applying the modified inputto a signal. And, the relevant details shall be described with referenceto FIGS. 2 to 10 later.

The bass controlling unit 400 generates an output signal by enhancingthe bass in a manner of applying an adaptive filter to the volumecontrol completed signal v(n). And, the relevant details shall bedescribed with reference to FIGS. 11 to 15 later.

FIG. 2 is a block diagram for details of the volume controlling unitshown in FIG. 1.

Referring to FIG. 2, the volume controlling unit 300 includes a gainsplitting part 320, non-linear gain modifying part 330 and a gainapplying part 340 and is able to further include a time-frequencyanalysis part 310 and a time-frequency synthesis part 350.

An input signal, an output signal and a gain of the volume controllingunit 300 have the following relation shown in Formula 1.

V[n]=gs[n]  [Formula 1]

In Formula 1, the v(n) indicates an output signal, the g indicates again, the s(n) indicates an input signal, the g indicates a specificgain (or a user gain input), and the n indicates a time index.

For simplification, assume that the gain g in Formula 1 is a maximumvolume if g=1 or a minimum volume (i.e., a zero volume) if g=0.

If the user gain (g) is a constant (i.e., a linear gain) instead of afunction per frequency band or the gain g is a small level value, alevel of a signal can be lowered equal to or smaller than an absolutehearing threshold for a low or high frequency band. This is because theabsolute hearing threshold of the low or high frequency band isrelatively greater than that of a middle band. If the level of thesignal is decreased equal to or lower than the absolute hearingthreshold, a listener may not be able to just hear the sound on thecorresponding frequency band at all. And, the relevant details shall bedescribed again later.

Therefore, the user gain g is not set to a constant but is split into anon-linear gain g₁ and a linear gain g₂. And, the relevant details shallbe explained with reference to the description of the gain splittingpart 320 later.

The time-frequency analysis part 310 generates a signal intime-frequency domain or a plurality of subband signals S[k, m] byperforming time-frequency conversion on the input signal s(n) in thetime domain. In this case, the time-frequency conversion can beperformed by SFT (short-time Fourier transform), by which the presentinvention is non-limited.

The gain splitting part 320 does not set the user input gain g to aconstant but splits the user gain input g into two factors (e.g., anon-linear gain g₁ and a linear gain g₂) occasionally. The relevantdefinition is represented as Formula 2.

g=g₁g₂  [Formula 2]

In Formula 2, the g indicates a user gain input, the g₁ is a non-lineargain, and the g₂ is a linear gain.

Meanwhile, the non-linear gain g₁ is a factor determined based on anabsolute hearing threshold P_(H). In determining the non-linear gain g₁and the linear gain g₂, it is able to use a high reference value and alow reference value. Details of a method of calculating a non-lineargain and a linear gain shall be described together with embodiments ofthe gain splitting unit 300 later.

The non-linear gain modifying part 330 generates a modified non-lineargain G₁[k, m] by modifying the non-linear gain g₁ using the absolutehearing threshold P_(H)[m] and power of input signal Ps[k, m]. In doingso, Formula 3 is usable.

$\begin{matrix}{{G_{1}\left\lbrack {k,m} \right\rbrack} = \sqrt{\frac{{g_{1}^{2}{\max \left( {{{P_{s}\left\lbrack {k,m} \right\rbrack} - {P_{H}\lbrack m\rbrack}},0} \right)}} + {P_{H}\lbrack m\rbrack}}{P_{S}\left\lbrack {k,m} \right\rbrack}}} & \left\lbrack {{Formula}\mspace{14mu} 3} \right\rbrack\end{matrix}$

In Formula 3, the G₁[k, m] indicates a modified non-linear gain, theP_(H)[m] indicates an absolute hearing threshold in a subband powerdomain, the g₁ indicates a non-linear gain, the Ps[k, m] indicates asubband power of an input signal, the k indicates a time index, and them indicates a frequency index.

The modified non-linear gain can be value invariant per frequency band.The power of the input signal Ps[k, m] may be estimated by thetime-frequency analysis part 310) based on input signal s[n], which doesnot put limitations on the present invention.

Meanwhile, FIG. 3 shows the example of the absolute hearing threshold,while FIG. 4 shows the example of the absolute hearing threshold in thesubband power domain. Particularly, FIG. 3 shows one example of theabsolute hearing threshold according to each sound pressure level (SPL)[dB] per frequency. In this case, the sound pressure level is a pressureof sound by logarithm unit with reference to a specific pressure (e.g.,20 micro Pascal) and its unit is decibel. Occasionally, sound can berepresented as sound intensity. Besides, a hearing sense actuallyperceived by a human for each sound pressure level is called soundloudness (unit: phon). And, an absolute hearing threshold indicates aminimum sound pressure level (SPL) that can be heard on each frequency.The minimum sound pressure level can correspond to a case that loudnessis 3 phons. And, FIG. 3 shows one example of the absolute hearingthreshold. Referring to FIG. 3, it can be observed that a sound pressurelevel corresponding to a hearing threshold is relatively high on a highfrequency band (e.g., over 10⁴ Hz) or a low frequency band (e.g., 0˜10²Hz). Hence, as mentioned in the foregoing description, when a volume islinearly adjusted, if the volume is adjusted into a low level, theintensity of the signal may fall into a level equal to or smaller thanan audible frequency. In particular, if the volume is linearly adjustedinto a low level, it may cause a problem that a human is unable to heara signal on a high or low frequency band. The present invention intendsto control a volume of a signal using a gain adaptive per frequencyband.

Meanwhile, the absolute hearing threshold is a value resulting fromaveraging values of sound pressure levels of minimum loudnessperceptible by a plurality of humans having different auditorystructures. In particular, since the absolute hearing threshold variesaccording to what kind of auditory structure is provided to a listener,the absolute hearing threshold may not be suitable according to who isthe listener. Further, if a location of an absolute-hearing listener iswell provided, a loudness model can have high precision. Therefore, if alocation of a listener changes, an audio apparatus is not able to obtainthe information on the change of the location.

Yet, the absolute hearing threshold represented as the sound pressurelevel per frequency band can be transformed into a subband power domain(or a power spectral domain). In case that the absolute hearingthreshold is transformed into the subband power domain, a specificelectric to acoustic power transfer function and approximate estimationof listening position can be explicitly or implicitly taken intoconsideration. In particular, when the absolute hearing threshold istransformed into a power domain from dB, if it is additionally scaled, apower spectral value approximately corresponds to a sound pressurelevel. One example of an absolute hearing threshold P_(H)[m] transformedinto a subband power domain is shown in FIG. 4. Since this algorithmputs limitation on a size of the absolute hearing threshold, it isadvantageous in that the threshold is prevented from being exceeded on avery low or high frequency. If this limit value P_(H) _(—) lim m is sethigher than a minimum value of the absolute hearing threshold P_(H)[m]by 20 dB at least, it can be represented as a dotted line part shown inFIG. 4.

Referring now to Formula 3, the g₁ is the formerly defined non-lineargain. And, a max function is provided to prevent a subband power valueof the input signal from becoming smaller than a value of the absolutehearing threshold (i.e., ‘P_(S)[k, m]−P_(H)[m]’ is prevented frombecoming negative). If the non-linear gain g₁ is set to 0, G₁[k, m]becomes a value that is not 1. If the non-linear gain g₁ is set to 1,the G₁[k, m] becomes 1.

Referring now to FIG. 2, the gain applying part 340 applies the modifiednon-linear gain G₁[k, m] generated by the non-linear gain modifying part330 and the linear gain g₂ generated by the gain splitting part 320 tothe input signal (i.e., a plurality of subband signals) S[k, m]. Thiscan be defined as Formula 4.

V[k,m]=g₂G₁[k,m]S[k,m]  [Formula 4]

In Formula 4, the V[k, m] indicates a per-subband output signal, the g₂indicates a linear gain, the G₁[k, m] indicates a modified non-lineargain, and the S[k, m] indicates a per-subband input signal.

In this case, the g₂ indicates a linear gain not varying per frequencyband and the G₁[k, m] indicates a non-linear gain generated on the basisof an absolute hearing threshold as a frequency function.

The time-frequency synthesis part 350 generates an output signal v(n) intime domain by performing frequency inverse transform on the per-subbandoutput signal. In this case, the output signal v(n) may be equal to theformer output signal in Formula 1.

In the following description, two embodiments of a process forgenerating a non-linear gain and a linear gain are explained in detailwith reference to FIGS. 5 to 10.

FIG. 5 is a detailed block diagram of a gain splitting unit shown inFIG. 2 according to a first embodiment of the present invention, andFIG. 6 is a flowchart for a gain generating method according to thefirst embodiment shown in FIG. 5. Although the gain splitting unit 320shown in FIG. 2 can include the detailed configuration shown in FIG. 5,it can include other components as well.

Referring to FIG. 5, the gain splitting unit 320 includes a comparingpart 324A, a non-linear gain generating part 326A and a linear gaingenerating part 328A and is able to further include a user gainreceiving part 322A. In the following description, a first embodiment320A of the gain splitting unit is explained with reference to FIG. 5and FIG. 6 together.

First of all, the user gain receiving part 322A receives a user gaininput (g) from a user interface, a memory or the like [S110]. In thiscase, as mentioned in the foregoing description of FIG. 1 and FIG. 2,the user gain input (g) can include data inputted by a user to adjust asize of an input signal or a value stored as a default value in a memoryor the like.

The comparing unit 324A receives a user gain g, a high reference valueg_(high) and a low reference value g_(low) [S120]. The comparing unit324A determines whether the user gain g is greater than the highreference value g_(high) and further determines whether the user gain gis greater than the low reference value g_(low) [S130 and S140]. In thiscase, a linear volume control is simply performed at the high referencevalue g_(high) or over an upper volume limit. Therefore, if the highreference value is determined as a high level, it is able to perform anon-linear gain control on the basis of an absolute hearing threshold ata high volume level. Meanwhile, the linear volume control is dominantlyperformed at the low reference value g_(low) or below an lower volumelimit. Therefore, once the low reference value g_(low) is set to 0, itis able to perform the non-linear control up to the interval of whichlevel becomes zero. In an interval between the high reference value andthe low reference value, the non-linear control on the basis of theabsolute hearing threshold is dominantly performed. The relevantrelation shall be explained in detail later.

If the user gain g is equal to or greater than the high reference valueg_(high) [‘No’ in the step S130], the comparing part 324A delivers theuser gain g to the linear gain generating part 328A only. If the usergain g is equal to or greater than the high reference value g_(high)[‘No’ in the step S130], the comparing part 324A delivers at least oneof the high reference value g_(high) and the low reference value g_(low)to each of the non-linear gain generating part 326A and the linear gaingenerating part 328A as well as the user gain g.

FIG. 7 shows one example for a case that a non-linear gain g₁ and alinear gain g₂ is determined according to whether a user gain g isgreater than or smaller than a high reference value g_(high) (and a lowreference value g_(low)). The following description is made withreference to FIG. 6 and FIG. 7.

As mentioned in the foregoing description, if the user gain g is equalto or greater than the high reference value g_(high) [‘No’ in the stepS130] [‘Interval 3’ in FIG. 7], the comparing part 324A delivers theuser gain g to the linear gain generating part 328A only. Hence, thenon-linear gain generating part 326A sets the non-linear gain g₁ to 1[S170], while the linear gain generating part 328A sets the linear gaing₂ equal to the user gain g (g₂=g) [S175].

On the contrary, if the user gain g is smaller than the high referencevalue g_(high) and is also equal to or greater than the low referencevalue g_(low) [‘No’ in the step S140] [‘Interval 2’ in FIG. 7], thenon-linear gain generating part 326A generates the non-linear gain g₁using the user gain and the high reference value g_(high) [S160], whilethe linear gain generating part 328A generates the linear gain g₂ usingthe high reference value g_(high) [S165].

Besides, if the user gain g is smaller than the low reference valueg_(low) [‘yes’ in the step S140] [‘interval 1’ in FIG. 7], thenon-linear gain generating part 326A generates the non-linear gain g₁using the high reference value g_(high) and the low reference valueg_(low) [S150], while the linear gain generating part 328A generates thelinear gain g₂ using the user gain g, the high reference value g_(high)and the low reference value g_(low) [S155].

One example of generating the non-linear gain and the linear gain isrepresented as Formula 5.

If g>g_(high), g₁=1.0, g₂=g  [Formula 5]

If g _(low) <g<g _(high) , g ₁ =g/g _(high) , g ₂ =g _(high)

If g<g _(low) , g ₁=g_(low) /g _(high) , g ₂ =g·g _(high) /g _(low)

In Formula 5, the g indicates a user gain input, the g_(high) indicatesa high reference value, the g_(low) indicates a low reference value, theg₁ indicates a non-linear gain, and the g₂ indicates a linear gain.

Meanwhile, FIG. 7 is a diagram for per-interval variations of linear andnon-linear gains. In a first case (‘interval 3’ in FIG. 7) of Formula 5,a non-linear gain is set to 1 and a user gain is handled as a lineargain. Hence, a volume control is linearly performed in this interval. Ina second case (‘interval 2’ in FIG. 7) of Formula 5, a non-linear gainincreases according to a size of a user gain g, whereas a linear gain isset to a constant (e.g., g_(high)). Hence, it can be observed that anon-linear volume control is dominant in this interval. In a third case(‘interval 1’ in FIG. 7) of Formula 5, a non-linear gain is set to aconstant (e.g., g_(low)/g_(high)) and a user gain is able to increaseaccording to a size of a user gain g. Hence, like the interval 3, it canbe observed that a linear volume control is dominant in this interval.

The non-linear gain factor g₁ and the linear gain factor g₂ aregenerated based on whether the user gain input is lower or higher thanat least one of a low reference value and a high reference value. Inparticular, when the user gain input is lower than a low reference value(in a third case (‘interval 1’ in FIG. 7) of Formula 5), the non-lineargain factor is generated as a fixed value, and the linear gain factor isgenerated using the user gain input. When the user gain input is equalto or higher than a low reference value (in a second case (‘interval 2’in FIG. 7) of Formula 5), the non-linear gain factor is generated usingthe user gain input and the linear gain factor is generated as a fixedvalue. The fixed value is determined according to the low referencevalue and a high reference value. When the user gain input is equal orhigher than a high reference value (in a first case (‘interval 3’ inFIG. 7) of Formula 5), the non-linear gain factor may be equal to oneand the linear gain factor may be equal to the user gain input.

As mentioned in the foregoing description regarding the absolute hearingthreshold, in case of adjustment into a high level, since the level toadjust is remote from the absolute hearing threshold, the adjustment islinearly controlled. On the contrary, in case of adjustment into a lowlevel, since a user-specific level can have a value similar to theabsolute hearing threshold, both linear and non-linear factors areadjusted. Therefore, it is able to prevent a specific frequency bandfrom decreasing below the absolute hearing threshold only.

So far, the first embodiment of the gain splitting unit is describedwith reference to FIGS. 5 to 7. In the following description, a secondembodiment 320B of the gain splitting unit is explained with referenceto FIGS. 8 to 10.

FIG. 8 is a detailed block diagram of a gain splitting unit shown inFIG. 2 according to a second embodiment of the present invention, andFIG. 9 is a flowchart for a gain generating method according to thesecond embodiment shown in FIG. 8. Unlike the first embodiment of thepresent invention, a second embodiment of the present invention relatesto a case that a non-linear gain control is performed in case of a usergain g greater than a low reference value g_(low) not based on whetherthe user gain g is higher than a high gain value g_(high). According tothe second embodiment of the present invention, since it may bedifficult to output a high level due to a speaker output capabilitylimit in case of a high level value, a volume is non-linearly controlledwith reference to an extent of human-perceptible loudness rather thancontrolling a volume linearly.

Referring to FIG. 8, a second embodiment 320B of a gain splitting unitincludes a comparing part 324B, a non-linear gain generating part 326Band a linear gain generating part 328B and is able to further include auser gain receiving part 322B.

First of all, the user gain receiving part 322B receives a user gaininput g like the former component of the same name in the firstembodiment [S210].

Likewise, the comparing part 32B receives a user gain g and a lowreference value g_(low) [S220] and then determines whether the user gaing is greater than the low reference value g_(low) [S240]. If the usergain g is smaller than the low reference value g_(low), the comparingpart 324B delivers the low reference value g_(low) to the non-lineargain generating part 326B and also delivers the user gain g and the lowreference value g_(low) to the linear gain generating part 328B. If theuser gain g is greater than the low reference value g_(low), thecomparing part 324B delivers the user gain g to the non-linear gaingenerating part 326B only.

A method of generating a non-linear gain and a linear gain per intervalis explained with reference to FIG. 9 and FIG. 10 as follows.

Referring to FIG. 9 and FIG. 10, if the user gain g is equal to orgreater than the low reference value g_(low) [‘No’ in the step S240 inFIG. 9] [‘Interval 2’ in FIG. 10], the non-linear gain g₁ is set equalto the user gain g [S260] and the linear gain g₂ is set to 0 [S265].

On the contrary, if the user gain g is smaller than the low referencevalue g_(low) [‘Yes’ in the step S240 shown in FIG. 9] [‘Interval 1’ inFIG. 10], the non-linear gain the non-linear gain g₁ is generated usingthe low reference value g_(low) [S250] and the linear gain g₂ isgenerated using the user gain g and the low reference value g_(low)[S225].

One example of generating the non-linear gain and the linear gainaccording to the second embodiment is represented as Formula 5.

If g>g_(low), g₁=g, g₂=1.0  [Formula 6]

If g<g _(low) , g ₁ =g _(low) , g ₂ =g/g _(low)

In Formula 6, the g indicates a user gain input, the g_(low) indicates alow reference value, the g₁ indicates a non-linear gain, and the g₂indicates a linear gain.

FIG. 10 is a diagram for per-interval variations of linear andnon-linear gains according to the second embodiment.

In a first case of Formula 6 (‘interval 2’ in FIG. 10), a user gain is anon-linear gain and a linear gain is set to 1. Hence, a volume controlis non-linearly performed in this interval. In a second case of Formula6 (‘interval 1’ in FIG. 10), a non-linear gain is a constant (e.g.,g_(low)) and a linear gain is able to increase according to a size of auser gain g. Hence, it can be observed that a linear control isrelatively dominant in this interval.

The non-linear gain factor g₁ and the linear gain factor g₂ aregenerated based on whether the user gain input is lower or higher thanat least one of a low reference value. In particular, when the user gaininput is lower than a low reference value (in a second case (‘interval1’ in FIG. 7) of Formula 6), the non-linear gain factor g₁ is generatedas a fixed value, and the linear gain factor g₂ is generated using theuser gain input. When the user gain input is equal to or higher than alow reference value (in a first case (‘interval 2’ in FIG. 7) of Formula6), the non-linear gain factor is generated using the user gain inputand the linear gain factor is generated as a fixed value. The fixedvalue is determined according to the low reference value. When the usergain input is equal or higher than a low reference value (in a firstcase (‘interval 2’ in FIG. 7) of Formula 6), the non-linear gain factormay be equal to the user gain input and the linear gain factor may beequal to one.

As mentioned in the foregoing description, if it is unable to output asignal at a high level due to a relatively low output of a speaker, asignal level outputtable from the speaker is limited despite that auser-specific volume is high. Therefore, it is able to perform anon-linear volume control based on an absolute hearing threshold.

Thus, the non-linear gain g₁ generated by the gain splitting part 320according to the first or second embodiment is inputted to the gainapplying part 240 shown in FIG. 2 and is then applied to an input signal(or subband signals for the input signal).

Stereo Channel and Multi-Channel

The aforesaid volume control method is applicable to a stereo signal andmulti-channel.

According to a first method, the volume control method is applied toeach audio channel (e.g., left channel, right channel, center channel,left surround channel, etc.), whereby linear and non-linear gains foreach of the channels are generated.

According to a second method, using a sum of subband powers P_(S)[k, m]for whole channels, a common gain G₁[k, m] applied to a stereo signal ora multi-channel is generated (As g₂ is signal-independent, it is equalto every channel). In calculating the G₁[k, m] according to Formula 3,the P_(S)[k, m] can insert a total of a sum corresponding to all audiochannels.

FIG. 11 is a detailed block diagram for an embodiment of a basscontrolling unit according to an embodiment of the present invention,and FIG. 12 is a flowchart of a method of controlling bass according toan embodiment of the present invention.

Referring to FIG. 11, a bass controlling unit 400 according to anembodiment of the present invention includes an indicator functionestimating part 420 and a filter obtaining part 430 and is able tofurther include a weight applying part 410 and a filter applying part440.

If a frequency is f(Hz), a loudspeaker has a frequency response of M(f).The frequency response M(f) indicates a real number for determining arelative strength indicating which sound is actually emitted via theloudspeaker per frequency band.

A filter h(t), which is an inverse number (i.e., 1/M(f)) of thefrequency response M(f), is applied to a signal before being outputtedvia the loudspeaker. Ideally, a frequency response of an acousticallyemitted sound is flat.

Meanwhile, in case of a small frequency response value, largeamplification is necessary for the corresponding frequency. Moreover,since distortion tends to be caused at a low level due to an excessivelyamplified low-frequency signal, a les volume-filtered signal can beintroduced into a loudspeaker only.

FIG. 13 is a graph for examples of a loudspeaker response and a targetequalization filter.

Referring to FIG. 13, one example of a frequency response M(f) of aloudspeaker is represented as a solid line. The frequency response ofthe loudspeaker starts to fall below a specific frequency f3, smoothlydecreases between frequencies f2 and f3, and then falls again below f1abruptly. To the loudspeaker having a poor frequency response of a lowfrequency, a considerable gain for a low frequency signal should begiven. FIG. 13 conceptionally shows a target equalization filter, i.e.,an inverse filter of the frequency response M(f). In an interval betweenfrequencies f1 and f2, for which an inverse filter for a frequencyresponse is suitable, the frequency response is inverted. Yet, in aninterval of which frequency is below f1, h(t) for the frequency responseis limited to a case that a frequency is f1 or the h(t) decreases. Ifthe filter h(t) for the frequency response is used, it is able toenhance the low frequency response of the loudspeaker. Yet, since lowfrequency energy is considerably boosted in a signal, if a volumeincreases higher, the loudspeaker can be distorted at a considerably lowsound level without equalization.

If the filter applied to the loudspeaker is adapted according to time,the bass can be extended as many as possible in a manner of avoidingoverdriving the loudspeaker.

A process for generating the filter is explained in detail withreference to FIG. 11 and FIG. 12 as follows.

First of all, a receiving part (not shown in the drawings) receives aninput signal [S310].

A weight applying part 410 calculates a power of a weighted signal usingan input signal s(t) and a weight filter w(t) [S320]. In this case, asignal power can be calculated according to the following formula. Inthis case, although the input signal s(t) may correspond to the volumecontrolled signal v(n) shown in FIG. 1, it may include an input signaly[n], a noise controlled signal or a normalized signal s(n).

P(t)=E{(w(t)*s(t))²}  [Formula 7]

In Formula 7, the P(t) indicates a signal power, the w(t) indicates aweight signal, the s(t) indicates an input signal, and the E{ }indicates a short-time average operator.

In this case, a single-pole average having a time constant of 200 ms isusable, by which the present invention is non-limited.

An indicator function estimating part 420 generates an indicatorfunction B(t) using the formerly generated signal power p(t) [S330].Using the signal power, the indicator function is generated inconsideration of a headroom (i.e., a difference between a distortiongenerated peak and a dynamic range of a corresponding signal) for aloudspeaker. In this case, the indicator function B(t) can be generatedby the following formula.

$\begin{matrix}{{B(t)} = {\min \left( {\frac{\max \left( {{{p(t)} - p_{\min}},0} \right)}{p_{\max} - p_{\min}},1.0} \right)}} & \left\lbrack {{Formula}\mspace{14mu} 8} \right\rbrack\end{matrix}$

In Formula 8, the B(t) indicates an indicator function, the Pmaxindicates a maximum power, and the Pmin indicates a minimum power.

FIG. 14 is a graph for an indicator function and a variation of anadaptive filter according to a signal power p(t).

Referring to (A) of FIG. 14, if a signal power p(t) is smaller than aminimum power Pmin, an indicator function B(t) is 0. As the signal powerp(t) increases, the indicator function B(t) gradually increases. If thesignal power p(t) becomes a maximum power Pmax, the indicator functionB(t) converges into 1.0.

A filter obtaining part 430 generates an adaptive filter using theindicator function B(t), a unity filter and an equalization filter[S340]. In doing so, Formula 9 is usable.

g(t)=B(t)δ(t)+(1−B(t))h(t)  [Formula 9]

In Formula 9, the g(t) indicates an adaptive filter, the B(t) indicatesan indicator function, the δ(t) indicates a unity filter (e.g., a deltaDirac function), and the h(t) indicates an equalization filter.

Referring to (B) of FIG. 14, a value of the adaptive filter g(t)according to the signal power p(t) and the indicator function B(t) isshown. If the signal power p(t) is equal to or smaller than a minimumpower, the adaptive filter g(t) directly becomes the equalization filterh(t). If the signal power p(t) is equal to or greater than a maximumpower, the adaptive filter g(t) becomes the unity filter δ(t). In aregion in-between (i.e., if smaller than the maximum power and greaterthan the minimum power), the adaptive filter has a configuration inwhich both of the equalization filter and the unity filter coexist.

FIG. 15 is a graph for an example of a loudspeaker response according toan indicator function.

Referring to FIG. 15, if an indicator function B(t) is 1 (i.e., theregion of (3) shown in FIG. 14), a frequency response of a loudspeakeris not modified (an adaptive filter corresponds to a unity filter).Meanwhile, if an indicator function B(t) is 0 (i.e., the region of (1)shown in FIG. 14), a frequency response of a loudspeaker is fullycorrected (an adaptive filter corresponds to an equalization filter). Ifan indicator function B(t) is greater than 0 and smaller than 1 (i.e.,the region of (2) shown in FIG. 14), a frequency response of aloudspeaker is partially corrected (an adaptive filter corresponds to acombination of a unity filter and an equalization filter).

<Frequency Domain Implementation>

The method for the bass controlling unit to control the bass, which wasdescribed with reference to FIG. 11 and FIG. 12, can be implemented intime-frequency domain as well as time domain. In particular, this methodis applicable to an input signal on which such frequency transform asfilter bank, shot time Fourier transform and the like is applied. Inthis case, a power p(t) of a weight-applied signal can be estimated froma subband signal. And, filter operation can be replaced bymultiplication in each subband (or frequency).

<Stereo Channel or Multi-Channel>

The method for the bass controlling unit to control the bass, which wasdescribed with reference to FIG. 11 and FIG. 12, is applicable to eachchannel signal of stereo- or multi-channel audio signal. Alternatively,an indicator unction B(t) (or a equalization boost parameter) can bejointly calculated for all signals using a power p(t) for a sum of allchannel. Likewise, an adaptive filter g(t), which is determined by B(t),is applied to al channels.

The audio signal processing apparatus according to the present inventionis available for various products to use. Theses products can be mainlygrouped into a stand alone group and a portable group. A TV, a monitor,a settop box and the like can be included in the stand alone group. And,a PMP, a mobile phone, a navigation system and the like can be includedin the portable group.

FIG. 16 is a schematic block diagram of a product in which an audiosignal processing apparatus according to one embodiment of the presentinvention is implemented. And, FIG. 17 is a diagram for explainingrelations between products in which an audio signal processing apparatusaccording to one embodiment of the present invention is implemented.

Referring to FIG. 16, a wire/wireless communication unit 510 receives abitstream via wire/wireless communication system. In particular, thewire/wireless communication unit 510 can include at least one of a wirecommunication unit 510A, an infrared unit 510B, a Bluetooth unit 510Cand a wireless LAN unit 510D.

A user authenticating unit 520 receives an input of user information andthen performs user authentication. The user authenticating unit 520 caninclude at least one of a fingerprint recognizing unit 520A, an irisrecognizing unit 520B, a face recognizing unit 520C and a voicerecognizing unit 520D. The fingerprint recognizing unit 520A, the irisrecognizing unit 520B, the face recognizing unit 520C and the speechrecognizing unit 520D receive fingerprint information, iris information,face contour information and voice information and then convert theminto user informations, respectively. Whether each of the userinformations matches pre-registered user data is determined to performthe user authentication.

An input unit 530 is an input device enabling a user to input variouskinds of commands and can include at least one of a keypad unit 530A, atouchpad unit 530B and a remote controller unit 530C, by which thepresent invention is non-limited.

A signal coding unit 540 performs encoding or decoding on an audiosignal and/or a video signal, which is received via the wire/wirelesscommunication unit 510, and then outputs an audio signal in time domain.The signal coding unit 540 includes an audio signal processing apparatus545. As mentioned in the foregoing description, the audio signalprocessing apparatus 545 corresponds to the above-described embodiment.Before an audio signal is outputted via the output unit, the audiosignal processing apparatus 545 performs at least one of noisecanceling, normalizing, volume control and bass control on the audiosignal. Thus, the audio signal processing apparatus 545 and the signalcoding unit including the same can be implemented by at least one ormore processors.

A control unit 550 receives input signals from input devices andcontrols all processes of the signal decoding unit 540 and an outputunit 560. In particular, the output unit 560 is an element configured tooutput an output signal generated by the signal decoding unit 540 andthe like and can include a speaker unit 560A and a display unit 560B. Ifthe output signal is an audio signal, it is outputted to a speaker. Ifthe output signal is a video signal, it is outputted via a display.

FIG. 17 is a diagram for the relation between a terminal and servercorresponding to the products shown in FIG. 16.

Referring to (A) of FIG. 17, it can be observed that a first terminal500.1 and a second terminal 500.2 can exchange data or bitstreamsbi-directionally with each other via the wire/wireless communicationunits. Referring to (B) of FIG. 17, it can be observed that a server 600and a first terminal 500.1 can perform wire/wireless communication witheach other.

An audio signal processing method according to the present invention canbe implemented into a computer-executable program and can be stored in acomputer-readable recording medium. And, multimedia data having a datastructure of the present invention can be stored in thecomputer-readable recording medium. The computer-readable media includeall kinds of recording devices in which data readable by a computersystem are stored. The computer-readable media include ROM, RAM, CD-ROM,magnetic tapes, floppy discs, optical data storage devices, and the likefor example and also include carrier-wave type implementations (e.g.,transmission via Internet). And, a bitstream generated by the abovementioned encoding method can be stored in the computer-readablerecording medium or can be transmitted via wire/wireless communicationnetwork.

Accordingly, the present invention is applicable to processing andoutputting of audio signals.

While the present invention has been described and illustrated hereinwith reference to the preferred embodiments thereof, it will be apparentto those skilled in the art that various modifications and variationscan be made therein without departing from the spirit and scope of theinvention. Thus, it is intended that the present invention covers themodifications and variations of this invention that come within thescope of the appended claims and their equivalents.

Accordingly, the present invention provides the following effects oradvantages.

First of all, for a volume level in the vicinity of an absolutethreshold of hearing, the present invention non-linearly controls avolume based on the absolute threshold of hearing, thereby preventingloudness of sound for a specific frequency band (e.g., a high frequency,a low frequency, etc.) from being inaudible or barely audible.

Secondly, in adjusting a volume into a level considerably higher than anabsolute threshold of hearing, the present invention applies anon-linear gain, thereby controlling the volume efficiently despite thatan output limit of a speaker is relatively low.

Thirdly, the present invention is able to distribute portions of linearand non-linear gains automatically according to whether a user gain issmaller than a reference value (e.g., high reference value and/or lowreference value).

Fourthly, the present invention automatically adjusts a portion formodifying a frequency response of a loudspeaker according to whether asignal power is big or small, thereby enhancing bass adaptively.

It will be apparent to those skilled in the art that variousmodifications and variations can be made in the present inventionwithout departing from the spirit or scope of the inventions. Thus, itis intended that the present invention covers the modifications andvariations of this invention provided they come within the scope of theappended claims and their equivalents.

1. A method for processing an audio signal, comprising: receiving, by anaudio processing apparatus, an input signal; estimating indicatorfunction using a signal power of the input signal; obtaining an adaptedfilter using the indicator function and an equalization filter; and,generating an output signal by applying the adapted filter to the inputsignal.
 2. The method of claim 1, wherein the indicator function is forheadroom for a loudspeaker.
 3. The method of claim 1, wherein theindicator function is estimated further using a maximum power and aminimum power.
 4. The method of claim 3, wherein: when the signal poweris greater than the maximum power, the adapted filter corresponds to aunity filter, when the signal power is equal to or smaller than themaximum power and greater than the minimum power, the adapted filtercorresponds to a combination of the unity filter and the equalizationfilter, and when the signal power is equal to or smaller than theminimum power, the adapted filter corresponds to the equalizationfilter.
 5. The method of claim 1, wherein the signal power is generatedusing a weight filter and the input signal.
 6. The method of claim 1,wherein the indicator function ranges from zero to one.
 7. An apparatusfor processing an audio signal, comprising: a receiving part receiving,by an audio processing apparatus, an input signal; an indicator functionestimating part estimating indicator function using a signal power ofthe input signal; a filter obtain part obtaining an adapted filter usingthe indicator function and an equalization filter; and, a filterapplying part generating an output signal by applying the adapted filterto the input signal.
 8. The apparatus of claim 7, wherein the indicatorfunction is estimated further using headroom for a loudspeaker.
 9. Theapparatus of claim 7, wherein the indicator function is estimatedfurther using a maximum power and a minimum power.
 10. The apparatus ofclaim 9, wherein: when the signal power is greater than the maximumpower, the adapted filter corresponds to a unity filter, when the signalpower is equal to or smaller than the maximum power and greater than theminimum power, the adapted filter corresponds to a combination of theunity filter and the equalization filter, and when the signal power isequal to or smaller than the minimum power, the adapted filtercorresponds to the equalization filter.
 11. The apparatus of claim 7,wherein the signal power is generated using a weight filter and theinput signal.
 12. The apparatus of claim 7, wherein the indicatorfunction ranges from zero to one.
 13. A computer-readable medium havinginstructions stored thereon, which, when executed by a processor, causesthe processor to perform operations, comprising: receiving, by an audioprocessing apparatus, an input signal; estimating indicator functionusing a signal power of the input signal; obtaining an adapted filterusing the indicator function and an equalization filter; and, generatingan output signal by applying the adapted filter to the input signal.